1. Field of the Invention
The present invention relates to high speed, high resolution signal transmission in data communication systems. In particular, the present invention is related to methods and system for providing high speed and high resolution data transmission for voice and/or data signals in V.90 modem telecommunication systems.
2. Description of the Related Art
Increasing demand for fast data connection from personal computers (PCs) for access from homes and businesses to data networks such as the world wide web through an Internet Service Provider (ISP) or a local area network (LAN) has given rise to the development in broadband networks employing Digital Subscriber Line (DSL) modems or cable modems. However, many existing locations in the United States and abroad currently do not have the necessary infrastructure to support such broadband networks. Thus, dial-up modem connection remains the best alternative to high speed access to data networks. Dial-up modem connections, however, have a significant limitation in the ability to transmit and/or receive data at a high transmission rate.
The introduction of V.90 standard 56K modems has increased the data transmission speed noticeably compared to the prior versions of modem technology such as V.34 modems. Prior to the introduction of the 56K modems, traditional modem standards assumed that both ends of a modem session have an analog connection to the public switched telephone network. Thus, data signals are converted from digital to analog and back again to the digital format such that the theoretical transmission rate was limited to approximately 33.6 Kbps. Moreover, transmission speeds above 33.6 kbps are not possible when the additional μ-law A/D conversion is introduced in the transmission path from the ISP server modem to the client modem required to accommodate the analog connection to the public switched telephone network. This limitation in speed is due to the quantization noise of the additional μ-law A/D conversion as well as the analog filters associated with the additional A/D converter and the corresponding μ-law D/A converter. It should be noted that the quantization noise of an A/D converter is a function of the resolution of the A/D conversion process. The resolution of the A/D conversion process is governed by the number of bits associated with each signal sample. The A/D converter used in typical telephony systems is an 8-bit companded converter, companding referring to the non-linear conversion characteristics of the A/D converter.
By contrast, the V.90 technology used in the 56 K modems approach data transmission from a different perspective. The V.90 technology assumes that one end of the modem session has a pure digital connection to the telephone network and takes advantage of this high speed digital connection. Indeed, by approaching the public switched telephone network as a digital network at the central office side, the V.90 technology can accelerate data transmission in the downstream path from the data network such as the internet to the remote side modem connected PCs. In this manner, the 56K modems differ from the prior modem technology such as the V.34 standard in that it digitally encodes downstream data instead of modulating it as analog modems as with V.34 modems.
More particularly, the presently available 56K modems achieve their high transmission speeds because the server modems provided by the ISP are digitally connected to the data network such as the public switched telephone network. As such, there is no A/D conversion in the path from the ISP server modem to the client modem at the remote end other than the A/D converter in the client modem itself. There is a D/A converter at the telephone company's Central Office (CO) equipment, but this D/A converter does not introduce quantization noise. Further detail on 56K modems and V.90 standard can be found in 3Com V.90 Technology, April, 1998 and in A Digital Modem and Analogue Modem Pair For Use On the Public Switched Telephone Network (PSTN) at Data Signaling Rates of Up to 56,000 bit/s Downstream and up to 33,600 bit/s Upstream, International Telecommunication Union, September, 1998.
FIG. 1 illustrates an eight line subscriber carrier system available from GoDigital Networks Corporation of Fremont, Calif., the assignee of the present invention. As shown, the subscriber carrier system 100 includes a Central Office Unit 101 that is coupled to a Customer Site Unit 102 by a bi-directional single twisted copper pair line 103. The Central Office Unit 101 is provided with eight Subscriber Line Interface Circuits (SLICs) 109, each of which is configured to emulate a telephone line termination. The Customer Site Unit 102 is provided with eight Subscriber Line Access Circuits (SLACs) 104, each of which are coupled to a subscriber line and further, configured to emulate the Central Office.
Moreover, the Central Office switch 110 located at the Central Office site receives an incoming digital transmitted at a rate of 56 Kbps from an ISP, for example. The Central Office switch 110, among other things, decodes the digital signal received from the ISP using a standard commercially available D/A converter 111 and transmits the decoded signal to the Central Office Unit 101 for each of the eight channels shown in FIG. 1.
As further shown in FIG. 1, each of the Central Office Unit 101 and the Customer Site Unit 102 are provided with a plurality of 8-bit μ-law 8 Ks/s codecs 105, 108 which are coupled to each of the SLICs 104 in the Central Office Unit 101, and to each of the SLACs 109 in the Customer Site Unit 102. Further shown in FIG. 1 are PCM buses which transmit and/or receive digital data bits from each of the 8-bit codecs 105, 108 in the Central Office Unit 101 and the Customer Site Unit 102, respectively, to the framing and transport mechanism 106, 107. The framing and transport mechanism 106, 107 of the Central Office Unit 101 and the Customer Site Unit 102, respectively are coupled to the single twisted copper pair 103.
The 8-bit codecs 104 in the Central Office Unit 101 are configured to encode analog data received from the Central Office switch (not shown) into a corresponding digital bit stream format and multiplex the same for transmission via the single twisted cable pair 103, while the 8-bit codecs 108 in the Customer Site Unit 102 are configured to demultiplex and decode the digital bit stream received from the Central Office Unit 101 into a corresponding analog form.
In particular, the μ-law codecs 105 receive 8,000 8-bit PCM code words per second, which translates to an aggregate bit rate of 64 Kbits/second, and converts the 8-bit PCM code words into corresponding analog voltage pulse signals each having 125 microsecond duration. The resulting output analog voltage pulse signals then has a stair-step characteristics. Moreover, the frequency spectrum of the analog voltage pulse signal is relatively broad compared to an analog voice band signal. For example, given a random sequence of PCM codewords, the frequency spectrum has a sinx/x shape, with a first spectral null at 8 KHz and repeated nulls at multiples of 8 KHz. It should be noted here that an analog voice band signal does not have appreciable energy above 4,000 Hz.
Furthermore, while not shown, there are provided low pass filters between each of the 8-bit codecs 105 and the SLICs 104 (SLACs 109 in the case of the Customer Site Unit 102) that pass energy up to approximately 3.4 KHz and attenuate the energy above 3.4 kHz such that energy above 4 KHz is severely attenuated. By way of example, the following Table 1 available from a datasheet for Lucent Technologies T7502 codec illustrates the attenuation characteristics for a given frequency:
TABLE 1Attenuation CharacteristicsFrequency (Hz)Typical attenuation<3,000    0 dB3,380 −.50 dB3,860−10.7 dB4,000−12.0 dB>4,600   −28 dB
As can be seen, the low pass filters are configured to contain the signal spectrum to a range of frequencies generally considered to be essential to human speech—approximately 300–3,400 Hz. Frequencies above 3,400 Hz are generally considered to have minimal impact on speech comprehension and furthermore, may cause crosstalk problems on subscriber lines.
It is to be noted that the mapping process of corresponding digital PCM codewords and analog voltages is non-uniform and confirms to μ-law encoding which was developed for processing speech signals where the step size between adjacent code words is proportional to the codeword magnitudes. This means that small magnitude code words are spaced very closely, and large magnitude codewords are spaced farther apart.
The μ-law encoding used in the United States —μ255 converter—requires the equivalent of a 13 bit linear DAC (i.e., 8,192 signal levels) to fully represent the 255 distinct levels (normally 256 levels for 8 bits, but there are two representations of “0”) because of non-uniform spacing. The signal-to-noise ratio (SNR) of a μ255 converter is substantially constant at 38–39 dB level over an input range of 30 dB or more. This characteristics permits the use of an 8-bit converter to produce acceptable voice encoding over a dynamic range that otherwise would require a 13-bit linear ADC.
The speech signal presented to the encoder may vary in input level from−10 to −40 dBm, but the signal presented to the far listener has the same SNR over this entire range of inputs. However, the μ-law encoding rule is not optimized for data communication over the POTS network.
Moreover, distortion is introduced by the low pass filters, the SLICs 104 and the subscriber loop 103. These distortions include amplitude and phase distortion (so-called linear distortions), and non-linear distortions such as limiting and clipping. Other sources of signal degradation include Gaussian and impulse noise, and crosstalk. The linear distortions are generally the dominant sources of signal degradation. These linear distortion sources cause an effect generally known as Inter-Symbol Interference (ISI) in communications systems. As the term implies, ISI makes it difficult for a subscriber modem to reconstruct the analog signal levels associated with digital codewords from μ-law DAC because at the ADC sampling rate in the above example, the ADC sample is a weighted sum of the most recently transmitted pulse plus a number of previously transmitted values.
Furthermore, the low pass filters introduce additional ISI to the signal. The ISI is not compensated prior to resampling at the 8-bit codecs 105. Indeed, the ISI can be large enough compared to the desired term to cause a translation to a different coding level than the original DAC level of the Central Office switch line card. As a result, the sequence of codewords after resampling is in general not identical to the sequence of codewords at the input to the Central Office DAC in this type of system.
The unwanted ISI term can be compensated in the modem receiver using known digital signal processing (DSP) adaptive equalization techniques. For example, the output of the DSP equalizer in a modem receiver is a sequence of levels that are essentially free of ISI. These levels can be converted back to their corresponding PCM codewords, and the result is a substantially perfect reconstruction of the sequence of PCM codewords that were provided to the μ-law DAC at the Central Office, on the assumption that noise, crosstalk, and other impairments are negligible. In practice, equalization is not perfect, and impairments are present to some extent, and PCM modems are only able to resolve among enough levels to facilitate transmission at 40 to 56 kbps.
As discussed above, in the subscriber carrier system 100 shown in FIG. 1, an extra A/D conversion process is required in the downstream path towards the subscribers in the Central Office Unit 101 as compared to the case where no pairgain system is present. For example, in the downstream path, the extra A/D conversion is necessary in the Central Office Unit 101 before the data stream is provided to the twisted copper pair 103. When a subscriber (or user) is connected to the Customer Site Unit 102 via a 56 Kbps modem, the extra A/D conversion required in the system described above reduces the subscriber's downstream modem speed by approximately 30% over the speed that can otherwise be achieved by the modem without the additional A/D conversion process.
In the V.90 modem transmission system, codeword sequence distortion is not introduced since there is no second encoding operation, and only approximately 14 dB of attenuation is introduced to the signal at 4,000 Hz. Moreover, a V.90 modem receiver at the Customer Site Unit 102 in the transmission system can equalize the ISI and recover the codeword sequence. In practice, however, noise, crosstalk, and residual ISI which the equalization algorithms were not able to remove contribute to signal degradation. Indeed, only a subset of the possible 255 distinct μ-law codewords are typically used, and the actual data rates obtained by V.90 modems are generally less than 56 kbps which is the highest potential speed for V.90 modems.
Furthermore, as described above, the pair gain system not only introduces significant codeword sequence distortion in the resampling process in the Central Office Unit, but also, the system low pass filters the signal three times with low pass filters with approximately 14 dB rolloff at 4,000 Hz. These filters are in the Central Office linecard, the Central Office Unit SLICs, and the Customer Site Unit SLACs. Thus, the total rolloff at 4,000 Hz is 42 dB or more by the time the signal reaches the modem at the customer premises. Thus, a V.90 modem receiver would have to compensate for this rolloff in order to properly reconstruct the sequence of codewords that are provided to the D/A converter at the Central Office switch if this reconstruction were possible. Of course, the reconstruction of the codeword sequence from the Central Office switch is not possible since it was distorted in the Central Office Unit's A/D converter. As a result, the effective data transmission rate will be low, possibly no better than that can be achieved with a V.34 modem.
In a pairgain system, the additional A/D conversion introduced in the transmission path is under the control of the pair gain system supplier. Moreover, the A/D converter in the pair gain system does not need to be an 8-bit companded type as generally used in the typical telephony systems. In fact, the A/D converter in a pair gain system can have a greater number of bits associated with each sample than the typical 8 bits. These extra bits (with linear encoding) will decrease the quantization noise and increase the attainable modem data transmission rates. However, a greater number of bits may be required to be carried over the pairgain system resulting in a reduction in the number of subscribers. Moreover, a pairgain system provider may also use sample rates higher than the traditional 8K samples per second (s/s) common to most existing telephony equipment. Higher sample rate and higher resolution will allow the pairgain system unit located near the Central Office to sample the telephone line without introducing significant distortion into the signal coming out of the Central Office D/A converter.
The A/D converter in the ISP server modem has a higher resolution than the Central Office D/A converter. The combination of higher resolution A/D converter and adaptive equalization algorithms in the digital signal processor (DSP) allows the modem to estimate the sequence of Central office D/A levels accurately, allowing a signaling alphabet of up to 128 codes (7 bits per symbol), or data rates of up to 56 kbps. It should be also noted that the higher resolution gained by the A/D converter is maintained at the terminal end of the pair gain system by a D/A converter of the same resolution.